Integrated internet telephony system and signaling method thereof

ABSTRACT

A VoIP network includes an integrated Internet telephony system having an all-in-one architecture in which an application layer gateway function, a signaling processing function and a media processing function are integrated. The integrated Internet telephony system performs ALG processing on a packet bound to a first address and processes a signaling message bound to a second address so as to set up a call session between an external SIP terminal connected to a public network and an internal SIP terminal connected to a private network, and performs media processing on an RTP packet based on a third address, the RTP packet exchanged through the call session. A SIP signaling gateway function, a media gateway function and SIP-ALG function are installed in one chip, and respective functions are allocated with different IP addresses by the application of a virtual interface and aliasing IP addresses.

CROSS-REFERENCE TO RELATED APPLICATION(S) AND CLAIM OF PRIORITY

This application makes reference to and claims all benefits accruingunder 35 U.S.C. §119 from an application for “INTEGRATED INTERNETTELEPHONY SYSTEM AND SIGNALING METHOD THEREOF” earlier filed in theKorean Intellectual Property Office on Jul. 23, 2007 and May 21, 2008and there duly assigned Serial No. 2007-73502 and 2008-47131.

TECHNICAL FIELD OF THE INVENTION

The present invention relates to an integrated Internet telephony systemand a signaling method thereof, particularly, which can realize, in onechip such as a central processing unit (CPU), a session initiationprotocol (SIP) signaling gateway function, a media gateway function andSIP-application layer gateway (SIP-ALG) function, which would otherwisebe realized in separate pieces of equipment in the prior art, and alsoallocate respective functions with different IP addresses by theapplication of a virtual interface and aliasing IP addresses, so as tofacilitate the construction of a Voice over Internet protocol (VoIP)network, minimize construction costs, and facilitate the maintenance,repair and management of the VoIP network.

BACKGROUND OF THE INVENTION

The Internet telephony or voice over Internet protocol (VoIP) is definedas communication technology that converts voice data into Internetprotocol (IP) data packets, which can be transmitted throughcommunication networks, in order to support voice conversation servicesnot only through a telephone network but also over the Internet.

A major advantage of the VoIP and Internet technology is that theyprovide a telephone service by utilizing the existing IP networks in anuntouched state, such that telephone users can be provided with longdistance and international telephone services in Internet and intranetenvironments while they pay local call rate.

The VoIP was introduced by major equipment providers, such as Cisco,VocalTec, 3Com and NetSpeak, in an attempt to promote the use of ITU-T,H.323 and the like, which are standards for transmitting voice or soundusing IP over the public Internet or through an intranet of a company.In order to promote the directory service standard, the VoIP forumallows users to locate other users. Furthermore, automatic calldistribution and the use of touch-phone signals for voice mails are alsoenabled.

As a characteristic feature, the VoIP uses real time protocol (RTP) inorder to support on-time arrival of packets in addition to its originalIP function. If a common public network is used, the characteristics ofbest-effort services make it difficult to support quality of service(QoS) for voice communication. As a result, the VoIP services can beprovided with higher quality when a private network managed by aseparate enterprise or an Internet telephone service provider (ITSP) isused.

The application layer gateway (ALG) is provided by an applicationgateway or a router. According to the ALG, a conventional firewall ornetwork address translator (NAT) can use several protocols in order toinspect packets transmitted between internal and external networks, anda verification process can be performed such that dynamically-allocatednetwork resources can pass through the firewall or NAT.

That is, the ALG is a device that routes a message packet, dynamicallyentering from an external network resource, to a specific host of asecurity-maintained internal network through inspection and verificationprocedures.

In particular, the session initiation protocol-ALG (SIP-ALG) istechnology designed to support communication between a SIP signalinggateway or a media gateway in an internal private network and a SIPproxy server or a SIP agent in an external public network.

FIG. 1 is an overview of the construction of a VoIP network thatprovides typical Internet telephony services.

As shown in FIG. 1, a private VoIP network using VoIP services generallyincludes an SIP signaling gateway 1, a media gateway 2, an internal SIPterminal 3 and an SIP-ALG router 4.

The SIP-ALG router 4 of the private VoIP network is connected to anexternal SIP proxy server 6 and external SIP terminals 7 and 8 throughthe Internet 5.

The SIP-ALG router 4 binds a SIP signaling message to a VoIP RTP streamthat the SIP signaling gateway 1 or the media gateway 2 transmits. TheSIP-ALG router 4 parses the bound packet, and then substitutes theprivate IP address of the packet with its own public IP address beforerouting the packet over the Internet.

Based on the ALG technology as described above, the SIP signalinggateway 1 and the media gateway 2 of the private VoIP network cancommunicate with the SIP proxy server 6 in the external network, whichuses a public IP address, by transmitting/receiving packets to/from theSIP proxy server 6.

However, as shown in FIG. 1, a typical private VoIP network isconstructed so that the SIP-ALG router 4, supporting the SIP-ALGfunction, is separated from the SIP signaling gateway 1 and the mediagateway 2.

In this construction separated from the SIP signaling gateway 1 and themedia gateway 2, the SIP-ALG router 4 overcomes the following problemsof the private VoIP network.

When any one of the SIP agents 1 to 3 in the private VoIP networkattempts to set up a session with an external user, following problemsmay occur. Firstly, the private IP address of the SIP agent 1 or 2,described in a SIP message or a Session Description Protocol (SDP),cannot be routed from outside. As a second problem, the NAT or thefirewall does not allow traffics from outside to pass through unlessthey are permitted. Finally, traffic cannot be transmitted from insideto outside unless it is allowed by the firewall.

Due to these problems, the SIP agents 1 and 2 and the SIP terminal 3inside the firewall cannot communicate with the external SIP proxyserver 6 or the external SIP terminal 7 or 8.

Accordingly, the SIP-ALG router 4 is designed to solve the NAT- andfirewall-related problems and to convert the private IP address intopublic IP address.

However, the SIP signaling gateway 1, the medial gateway 2 and therouter 4 supporting the SIP-ALG functions are required to be separatelypurchased in order to operate the foregoing private VoIP network. Ofcourse, a company or firm operating the private VoIP network has toseparately maintain, repair and manage respective pieces of equipment.

Accordingly, it is difficult for a network operator, who has to manageseveral pieces of equipment, to easily manage the private VoIP network.

In the meantime, an integrated Internet telephony system having anall-in-one architecture in which SIP signaling, media gateway andSIP-ALG functions are integrated can be provided by realizing the SIPsignaling gateway 1, the media gateway 2 and the SIP-ALG router 4 in onepiece of equipment in order to facilitate the maintenance, repair andmanagement of the equipment while minimizing construction costs of theprivate VoIP network.

Also, there is a problem in that the SIP signaling gateway 1, the mediagateway 2 and the SIP-ALG router 4 are required to be allocated withdifferent unique IP addresses while the integrated Internet telephonysystem uses only one IP address.

SUMMARY OF THE INVENTION

To address the above-discussed deficiencies of the prior art, it is aprimary object of the present invention to solve the foregoing problemswith the prior art, and therefore the present invention is directed toan integrated Internet telephony system and a signaling method thereof,which can simultaneously process, in one piece of equipment, a pluralityof voice of Internet protocol (VoIP) -related functions, such as asession initiation protocol (SIP) signaling gateway, media gateway andsession initiation protocol-application layer gateway (SIP-ALG)functions, which would otherwise be operated in separate pieces ofequipment according to the prior art.

The present invention is also directed to an integrated Internettelephony system and a signaling method thereof, which can allocaterespective parts such as a SIP signaling gateway, a media gateway and aSIP-ALG with different IP addresses in order to provide an Internettelephone service.

The present invention is further directed to an integrated Internettelephony system and a signaling method thereof, in which the SIP-ALGmanages different address information of the SIP signaling gateway andthe media gateway, such that a signaling packet and a real time protocol(RTP) packet, necessary for an Internet telephone service, can betransmitted to/received from the SIP signaling gateway and the mediagateway.

According to an aspect of the invention, there is provided a voice ofInternet protocol (VoIP) network comprising an integrated Internettelephony system having an all-in-one architecture in which anapplication layer gateway function, a signaling processing function anda media processing function are integrated. The integrated Internettelephony system performs application layer gateway (ALG) processing ona packet bound to a first address and processes a signaling messagebound to a second address so as to set up a call session between anexternal session initiation protocol (SIP) terminal connected to apublic network and an internal SIP terminal connected to a privatenetwork, and performs media processing on a real time protocol (RTP)packet based on a third address, the RTP packet exchanged through thecall session.

The integrated Internet telephony system may include a first interfaceconnected to the public network; a second interface connected to theprivate network; and a third interface connected to a different networkfrom the second interface.

In the VoIP network, the first address is a private Internet protocol(IP) address of the second interface, the second address is in the samenetwork as the first address, but is a different logical IP address fromthe first address, and the third address is in the same network as thethird interface, but is a different logical IP address from the thirdinterface.

In the VoIP network, the second address is an IP address aliased fromthe first address, and the third address is an IP address aliased fromthe address of the third interface.

The integrated Internet telephony system may modify private or publicaddress information in a header field of the signaling message, which isbound to the first address, by mapping public or private addressinformation; and may modify destination address information of thesignaling message, which is bound to the first address, into the secondaddress.

Further, the integrated Internet telephony system may manage addressinformation of the internal SIP terminal, included in a sessiondescription protocol (SDP) part of the signaling message, using aninbound mapping table when the signaling message is going out from theprivate network to the public network; and may manage addressinformation of the external SIP terminal included in the SDP part of thesignaling message using an outbound mapping table when the signalingmessage is an in-coming message.

In addition, the integrated Internet telephony system may modifydestination address information of an RTP packet by referring to theinbound mapping table when the RTP packet is an out-going packet, andmodify the destination address information of the RTP packet byreferring to the outbound mapping table when the RTP packet is anin-coming packet.

When the call session is set up, the internal SIP terminal or theexternal SIP terminal may exchange the RTP packet based on thedestination address information included in the SDP part of thesignaling message.

According to another aspect of the invention, there is provided anintegrated Internet telephony system, which provides an Internettelephone service through a voice of Internet protocol (VoIP) network.The integrated Internet telephony system includes a local area network(LAN) interface connected to a private VoIP network using a firstprivate Internet protocol (IP) address; a wide area network (WAN)interface connected to a public network using a public IP address; avirtual interface having a virtual private IP address; a sessioninitiation protocol (SIP) signaling gateway module processing asignaling message, which is bound to a second private IP address, so asto set up a call session; a media gateway module connected to thevirtual interface through a third private IP address so as to process areal time protocol (RTP) packet; and a session initiationprotocol-application layer gateway (SIP-ALG) module modifyingdestination address information of the bounded signaling message andaddress information included in a session description protocol (SDP)part of the bounded signaling message.

The SIP-ALG module may modify destination address information of anout-going signaling message and private address information included inthe SDP part of the out-going signaling message into public addressinformation; and may modify destination address information of anin-coming signaling message into the first private IP address, andmodifies public address information of an external SIP terminal includedin the SDP part into private address information.

Further, the SIP-ALG module may include a SIP parser parsing the boundsignaling message; a field modification module modifying addressinformation in an IP header of the signaling message; an SDP processingmodule converting address information in the SDP part of the signalingmessage; and a SIP mapping table storing address information mapped toaddress information included in the signaling message when the signalingmessage is an in-coming or out-going message.

In addition, the SIP-ALG module may modify the destination addressinformation of the signaling message into the first private IP addresswhen the signaling message is an in-coming message, and modifies sourceaddress information of the in-coming signaling message into the firstprivate IP address; may set the public IP address as destination addressinformation in the SDP part of the signaling message; and may modifysource address information of the signaling message into the public IPaddress when the signaling message is an out-going message, and modifiesaddress information in the SDP part into an external IP address of anexternal terminal.

According to a further aspect of the invention, there is provided asignaling method of an integrated Internet telephony system, which hasan all-in-one architecture in which a session initiationprotocol-application layer gateway (SIP-ALG) function, a sessioninitiation protocol (SIP) gateway function and a media processingfunction are integrated, and which includes a first interface connectedto a private voice of Internet protocol (VoIP) network and a secondinterface connected to a public network. The method includes proceduresof: adding, at the integrated Internet telephony system, a virtualinterface belong to a different network from a first address; managing,at a first module in charge of the SIP-ALG function, address informationincluded in a signaling message, which is bound to the first address,using a mapping table; modifying, at the first module, addressinformation of the signaling message and address information included ina session description protocol (SDP) part of the signaling message;setting up, at a second module in charge of the SIP signaling gatewayfunction, a call session based on the signaling message, which is boundto a second address; and exchanging, at an internal SIP terminal in theprivate VoIP network and an external terminal in the public network, areal time protocol (RTP) packet using a third address.

The signaling method may further include a procedure of performing, at athird module in charge of the media processing function, mediaprocessing on the RTP packet using the third address, which is connectedto the virtual interface.

The procedure of managing, at the first module in charge of the SIP-ALGfunction, address information included in a signaling message, which isbound to the first address, using a mapping table, may includeprocedures of: managing private address information of the internal SIPterminal in the SDP part of the signaling message, which is going outfrom the private VoIP network to the public network, using an inboundmapping table; and managing public address information of the externalSIP terminal in the SDP part of the signaling message, which is comingin, using an outbound mapping table.

The procedure of modifying, at the first module, address information ofthe signaling message and address information included in an SDP of thesignaling message, may include procedures of: modifying the addressinformation of the signaling message into corresponding private IPaddress information or public IP address information by referring to arespective one of the mapping tables; modifying destination sourceaddress information and source address information of the signalingmessage, which is coming in, from the public IP address into the firstprivate IP address and from address information of the external SIPterminal into the first private IP address by referring to the inboundmapping table, and transmitting the signaling message to the internalSIP terminal; and modifying source address information of the SIPsignaling message and source information in a payload field of the SIPsignaling message, which is going out, into the public IP addressinformation and external IP address of the external SIP terminal byreferring to the outbound mapping table.

As set forth above, the invention can simultaneously realize, in onepiece of equipment, a plurality of VoIP-related functions, such as a SIPsignaling gateway function, a media gateway function and a SIP-ALGfunction, which would otherwise be operated in separate pieces ofequipment according to the prior art. This, as a result, can facilitatethe construction of a VoIP network, which provides an Internet telephoneservice, minimize construction costs of the VoIP network, and facilitatethe maintenance, repair and management of the VoIP network.

Further, the SIP signaling gateway function, the media gateway functionand SIP-ALG function are installed in one chip, such as centralprocessing unit (CPU), and are allocated with different IP addresses bythe application of a virtual interface and aliasing of IP addresses,such that a call session for an Internet telephone service between theSIP terminals can be set up and RTP packets can be exchanged even thoughthe SIP signaling gateway function, the media gateway function andSIP-ALG function are all-in-one integrated.

Moreover, address information of a signaling message necessary for anInternet telephone service and an IP address included in an SDP part canbe so managed that a call session can be set up between an internal SIPterminal in a private VoIP network such as a firewall and an externalSIP terminal in a public network such as the Internet.

Before undertaking the DETAILED DESCRIPTION OF THE INVENTION below, itmay be advantageous to set forth definitions of certain words andphrases used throughout this patent document: the terms “include” and“comprise,” as well as derivatives thereof, mean inclusion withoutlimitation; the term “or,” is inclusive, meaning and/or; the phrases“associated with” and “associated therewith,” as well as derivativesthereof, may mean to include, be included within, interconnect with,contain, be contained within, connect to or with, couple to or with, becommunicable with, cooperate with, interleave, juxtapose, be proximateto, be bound to or with, have, have a property of, or the like.Definitions for certain words and phrases are provided throughout thispatent document, those of ordinary skill in the art should understandthat in many, if not most instances, such definitions apply to prior, aswell as future uses of such defined words and phrases.

BRIEF DESCRIPTION OF THE DRAWINGS

For a more complete understanding of the present disclosure and itsadvantages, reference is now made to the following description taken inconjunction with the accompanying drawings, in which like referencenumerals represent like parts:

FIG. 1 is an overview illustrating the construction of a VoIP networkthat provides typical Internet telephony services;

FIG. 2 is an overview illustrating the construction of a VoIP networkincluding an integrated Internet telephony system, to which theinvention is applicable;

FIG. 3 is an overview illustrating the construction of a VoIP networkincluding an integrated Internet telephony system according to anembodiment of the invention;

FIG. 4 is a block diagram illustrating an exemplary embodiment of theintegrated Internet telephony system according to an embodiment of theinvention;

FIG. 5 is a block diagram illustrating a function of the SIP-ALG moduleof the integrated Internet telephony system shown in FIG. 4;

FIG. 6 is a flow diagram illustrating a method of processing SIPsignaling messages in the integrated Internet telephony system accordingto an embodiment of the invention;

FIGS. 7A and 7B are SIP mapping tables of the integrated Internettelephony system according to an embodiment of the invention, in whichFIG. 7A shows an inbound mapping table and FIG. 7B shows an outboundmapping table;

FIG. 8 illustrates a method in which a SIP-ALG module of an embodimentof the invention manages the outbound mapping table and the inboundmapping table; and

FIGS. 9A to 9F are conceptual views illustrating SIP signaling messageflows in the integrated Internet telephony system according to anembodiment of the invention.

DETAILED DESCRIPTION OF THE INVENTION

FIGS. 1 through 9F, discussed below, and the various embodiments used todescribe the principles of the present disclosure in this patentdocument are by way of illustration only and should not be construed inany way to limit the scope of the disclosure. Those skilled in the artwill understand that the principles of the present disclosure may beimplemented in any suitably arranged telephony system.

Hereinafter, an integrated Internet telephony system and a signalingmethod thereof according to the present invention will be described morefully with reference to the accompanying drawings, in which exemplaryembodiments thereof are shown. In the following description, somefunctions or constructions will not be described in detail since theywould obscure the invention in unnecessary detail.

In the following detailed description of the invention, a primary IPaddress, which is allocated to a piece of network equipment and is usedfor a SIP-ALG function, will be referred to as the “first address,” andan address, which is aliased from the primary IP address and is used fora SIP signaling gateway function, will be referred to as the “secondaddress.” Further, an address, which is aliased from an IP address of avirtual interface and is used in a media gateway module, will bereferred to as the “third address.”

The exemplary embodiments of the invention will be described withrespect to a case where an RTP packet exchanged through a call sessionincludes voice data, but this is not intended to limit the invention.Rather, it can be equally applied to a case where the RTP packetincludes other data, such as image data.

FIG. 2 is an overview of the construction of a VoIP network including anintegrated Internet telephony system, to which the invention isapplicable.

The private VoIP network shown in FIG. 2 includes first and seconddigital telephone terminals 11 and 12, first and second internal SIPterminals 13 and 14, a call server 15 and an integrated Internettelephony system 16.

Herein, those agents such as the digital telephone terminals 11 and 12and the internal SIP terminals 13 and 14, which are provided in theinternal network of the private VoIP network in order to set up atelephony session for providing an Internet telephony service andthereby exchange an RTP packet including voice data, will be referred toas “internal SIP terminal,” and those agents connected to the Internetwill be referred to as “external SIP terminal.”

The integrated Internet telephony system 16 has an all-in-onearchitecture in which those components of the typical VoIP network shownin FIG. 1, such as the SIP signaling gate 1, the media gateway 2 and theSIP-ALG router 4, operate on one chip (CPU).

The integrated Internet telephony system 16 shown in FIG. 2 isequipment, which is obtained by integrating those components shown inFIG. 1, such as the SIP signaling gateway 1, the media gateway 2 and theSIP-ALG router 4, without using a virtual interface 140 (see FIG. 4) ofthe invention.

The integrated Internet telephony system 16 as described above uses asingle primary IP address. Since 5060 ports used by the SIP-ALG and theSIP signaling cannot be simultaneously bound to one IP address, both theSIP signaling function and the SIP-ALG function, which have to usedifferent IP addresses, cannot be simultaneously enabled.

That is, the SIP signaling function and the SIP-ALG function have to usedifferent unique IP addresses. As shown in FIG. 2, when the SIPsignaling function and the SIP-ALG function are simply integrated intoone piece of equipment, the integrated Internet telephony system can useonly one IP address, so that both the SIP signaling function and theSIP-ALG function, which are required to use different IP addresses,cannot be simultaneously enabled.

Due to this problem, the private IP address of any one of the internalSIP terminals 13 to 14 or a call server 15 is inserted into the IPaddress field of a SIP message that the call server 15 transmits to anexternal SIP terminal 17. The private IP address of the call server 15is inserted into the source IP address field of an RTP packet includingvoice data that the call server 15 transmits to the external SIPterminal 17.

In this case, the external SIP terminal 17 generates an SIP signalingmessage or an RTP packet by using the private IP address of the internalSIP terminal 13 to 14 or the call server 15, inserted in the SIP messageor the RPT packet, as a destination address. However, since the privateIP address cannot be routed through the Internet 103, the SIP signalingmessage or the RTP packet from the external SIP terminal 17 cannot besent to the SIP signaling gateway or the internal SIP terminal 13 to 14,which take charge of the SIP signaling of the integrated Internettelephony system of the private VoIP network.

Similarly, supposing that the second internal SIP terminal 14 transmitsa SIP register message, which requests registration from the externalSIP server 18, a source IP address field in the SIP register messagealso has the private IP address of the second internal SIP terminal 14.

Since the destination IP address of the SIP register message is thepublic IP address of the external SIP server 18, the SIP registermessage is normally transmitted from the second internal SIP terminal 14to the external SIP server 18.

However, since the external SIP server 18 is notified of merely theprivate IP address of the second SIP terminal 14, a SIP registerresponse message and an RTP packet in response to the SIP registermessage according to the SIP register message cannot be sent to theprivate VoIP network.

Both the SIP signaling function and the SIP-ALG function cannotsimultaneously operate in the integrated Internet telephony system sincea SIP-ALG module 130 (see FIG. 4) of the integrated Internet telephonysystem 16 operates in the user datagram protocol (UDP) 5060 port, towhich a SIP signaling gateway module 120 (see FIG. 4) of the integratedInternet telephony system 16, in charge of a SIP signaling function,binds a signaling message.

That is, one SIP standard port 5060 cannot be used by two applicationsoftware modules (including the SIP signaling gateway module 120 and theSIP-ALG module 130).

As a result, the integrated Internet telephony system 16 of FIG. 2cannot simultaneously perform the SIP-ALG function, the SIP signalinggateway function and the media gateway function.

Below, a description will be given of an integrated Internet telephonysystem of the invention, which can perform the SIP signaling gatewayfunction, the media gateway function and the SIP-ALG function.

FIG. 3 is an overview of the construction of a VoIP network including anintegrated Internet telephony system of the invention.

As shown in FIG. 3, the VoIP network includes internal SIP terminals (aninternal digital telephone terminal 101 and an internal SIP terminal102), an integrated Internet telephony system 100 and external SIPterminals (an external SIP proxy server 104 and an external SIP terminal105) connected to the integrated Internet telephony system 100 via theInternet 103.

Referring to FIG. 3, supposing that a subnet mask be 255.255.255.0, theintegrated Internet telephony system 100 has a public IP address (e.g.,61.77.100.20) and a private IP address (LAN IP address) (e.g.,10.0.0.1.).

Further, the external SIP proxy server 104 has a public IP address61.77.100.30, and the external SIP terminal 105 has a public IP address61.77.100.40.

The integrated Internet telephony system 100 is an integrated systemthat can simultaneously carry out not only the SIP-ALG function but alsothe SIP signaling gateway and media gateway functions.

Firstly, of several modules of the integrated Internet telephony system100, the SIP-ALG module 130 (see FIG. 4) using a primary IP address(hereinafter, referred to as “first IP address”) parses SIP signalingmessages, which are sent from inside to outside of a private VoIPnetwork or vice versa, and reconstructs the parsed messages (addressmodification and SDP part correction), in such a manner that themessages can be recognized by the external/internal SIP terminals (theinternal digital telephone terminal, the internal SIP terminal, the SIPproxy server and the external SIP terminal), before transmitting themessages.

That is, the SIP-ALG module 130 parses a SIP signaling message such as aSIP request message or a SIP response message and modifies the addressinformation (IP or port information) of the SIP request message or theSIP request message, such that the internal SIP terminal inside theprivate VoIP network (or the firewall) can recognize the message.

The SIP-ALG module 130 corrects a session description protocol (SDP)part of a voice data, which is transmitted on an RTP packet, andmaintains and manages related information, such that the voice data canactually be sent to the internal SIP terminal 101 or 102.

The integrated Internet telephony system 100 of FIG. 3, which isproposed by the invention in order to overcome the problem occurring inthe integrated Internet telephony system of FIG. 2, enables the SIP-ALGmodule 130 (see FIG. 4), the SIP signaling gateway module 120 (see FIG.4) and a media gateway module 150 (see FIG. 4) to use different addressinformation by using the virtual interface 140 (see FIG. 4) and alias IPaddress technology.

Describing in more detail, the primary IP address allocated to the CPUof the integrated Internet telephony system 100 is set (as a firstaddress) to be used by the SIP-ALG module 130, and a virtual IP address(hereinafter, referred to as “second IP address”) is allocated by analias method so as to be used by the SIP signaling gateway module 120.

In addition to a local area network (LAN) interface 110 connected to theinternal network (private VoIP network) and a wide area network (WAN)interface 160 connected to the Internet 103, the integrated Internettelephony system 100 further includes the virtual interface 140, throughwhich the media gateway module 150 processing RTP packets can exchangethe RTP packets.

The integrated Internet telephony system 100 also enables the mediagateway module 150 to use a third address, which is aliased from theaddress information of the virtual interface 140.

Thus, the SIP-ALG module 130 binds a SIP signaling message to the firstaddress, the SIP signaling gateway module 120 binds a SIP signalingmessage to the second address, and the media gateway module 150exchanges an RTP packet using the third address.

In the integrated Internet telephony system 100 proposed by theinvention, the SIP-ALG module 130, the SIP signaling gateway module 120and the media gateway module 150 use different address information, andthus they can operate even if they are installed in one CPU of theintegrated system.

While a description is given of a case in which unique IP addresses,different from the IP address of the SIP-ALG module 130, are allocated(given) respectively to the SIP signaling gateway module 120 and themedia gateway module 150 by an alias addressing method in such a mannerthat the SIP-ALG module 130, the SIP signaling gateway module 120 andthe media gateway module 150 use different IP addresses, different IPaddresses can be allocated respectively to the SIP-ALG module 130, theSIP signaling gateway module 120 and the media gateway module 150 by adifferent address allocation method.

For the operation of the media gateway module 150, the integratedInternet telephony system 100 additionally designates a virtualinterface. After that, the media gateway module 150 uses an alias IPaddress belonging to the same network as the virtual interface 140(hereinafter, referred to as “third address”). That is, the mediagateway module 150 processes an RTP packet using the third address,which is aliased from the IP address allocated to the virtual interface140.

As a result, the integrated Internet telephony system 100 can performall the operations of the SIP signaling gateway 1, the media gateway 2and the SIP-ALG router 4 shown in FIG. 1 using the virtual interface 140and the added alias IP address. As such, the respective pieces ofequipment for providing an Internet telephony service can be implementedas one integrated piece of equipment, to thereby facilitate theconstruction of a VoIP network, minimize construction costs of the VoIPnetwork, and facilitate the maintenance, repair and management of theVoIP network.

Below, a description will be given of a method of adding the virtualinterface 140 and a method of allocating an alias IP address in theintegrated Internet telephone system 100 of the invention.

FIG. 4 is a block diagram illustrating an embodiment of the integratedInternet telephony system according to the invention.

Referring to FIG. 4, the integrated Internet telephony system 100 isrealized on one chip (CPU), in which the SIP-ALG module 130, the SIPsignaling gateway module 120 and the media gateway module 150 areall-in-one integrated. Here, the SIP-ALG module 130 manages a SIP-ALGfunction, the SIP signaling gateway module 120 processes a SIP signalingmessage to set up a call session for an Internet telephone service, andthe media gateway module 150 processes an RTP packet including voicedata.

As such, the SIP-ALG module 130, the SIP signaling gateway module 120and the media gateway module 150 can be realized as software functionblocks on one chip.

Further, the integrated Internet telephony system 100 includes the LANinterface 110, the WAN interface 160 and the virtual interface 140.Here, the LAN interface 110 manages data transmission to/from theprivate VoIP network, the WAN interface 160 manages to transmit andreceive data (e.g., SIP signaling messages and RTP packets) through theInternet, and the virtual interface module 140 is added to allow themedia gateway module 150 to transmit and receive the RTP packets.

Below, a description will be given of IP addresses allocated torespective interfaces 110, 140 and 160.

As shown in FIG. 4, the IP address (public IP address) of the WANinterface 160 is 61.77.100.20. Since the WAN interface 160 interworkswith the Internet 103 as a public network, the address 61.77.100.20 is apublic IP address.

The IP address of the LAN interface 110 is 10.0.0.1. Since the LANinterface 110 interworks with a private network, such as a private VoIPnetwork, the IP address 10.0.0.1 is a private IP address.

The public IP address and the private IP address of the integratedInternet telephony system 100 correspond respectively to the IP addressof the WAN interface 160 and the IP address of the LAN interface 110.Thus, the public IP address of the integrated Internet telephony system100 is 61.77.100.20, and the private IP address of the integratedInternet telephony system 100 is 10.0.0.1.

The integrated Internet telephony system 100 further includes thevirtual interface 140, which has a private IP address different from theprivate IP address of the LAN interface 110. For example, the private IPaddress of the virtual interface 140 can be 10.0.1.1.

Supposing the subnet mask of the virtual interface 140 be 255.255.255.0,the virtual interface 140 is an interface that belongs to a differentnetwork from the WAN interface 160 or the LAN interface 110.

The media gateway module 150 can transmit and receive an RTP packetthrough the virtual interface 140 only when it is in the same network asthe virtual interface 140. Accordingly, the media gateway module 150uses the third address (e.g., 10.0.1.2) aliased from the IP address ofthe virtual interface 140.

The private IP address of the SIP signaling gateway module 120 is10.0.0.2, which is aliased, for example, from the private IP address ofthe integrated Internet telephony system 100, particularly, the privateIP address of the LAN interface 110.

The integrated Internet telephony system 100 has not only the primary IPaddress 10.0.0.1 but also 10.0.0.2 and 10.0.1.1, which are the privateIP address of the SIP signaling gateway module 120 and the private IPaddress of the virtual interface 140, respectively, in which the SIPsignaling gateway module 120 uses the private IP address aliased fromthe private IP address that the SIP-ALG module 130 uses (i.e., thesecond address), and the media gateway module 150 uses the private IPaddress aliased from the private IP address of the virtual interface 140(i.e., the third address).

According to the invention, one physical piece of equipment (theintegrated Internet telephony system 100) can have a plurality oflogical IP addresses (e.g., 10.0.0.2 and 10.0.1.2), and an alias IPaddress method is applied as an example of adding a plurality of logicalIP addresses.

The SIP signaling gateway module 120 uses the IP address 10.0.0.2, whichis aliased from the private IP address 10.0.0.1 of the LAN interface110, and the SIP signaling gateway module 120 can set the private IPaddress 10.0.0.1 of the LAN interface 110 as a default gateway address.

As such, when the default gateway address of the SIP signaling gatewaymodule 120 is set as the private IP address of the LAN interface 110, aSIP signaling message generated by the SIP signaling gateway module 120is forwarded to the LAN interface 110, and is then routed to an externalSIP terminal through the SIP-ALG module 130 and the WAN interface 160.

The integrated Internet telephony system 100 additionally sets thevirtual interface 140 in addition to the LAN interface 110 for thepurpose of the operation of the media gateway module 150.

The integrated Internet telephony system 100 of the embodiment shown inFIG. 4 illustrates an exemplary case in which the virtual interface 140having the private IP address 10.0.1.1 is added, and the media gatewaymodule 150 uses the private IP address 10.0.1.2 aliased from the privateIP address of the added virtual interface 140.

The SIP-ALG module 130 processes the ALG function by binding the SIPsignaling message through the public IP address 61.77.100.20 and theprivate IP address 10.0.0.1 of the integrated Internet telephony system100.

In order to bind the SIP signaling message, the SIP-ALG module 130monitors the 5060 port (i.e., the port of the SIP signaling message,which is generally defined in the SIP).

Further, the SIP-ALG module 130 acquires all the private IP addressesused in the integrated Internet telephony system 100, such as theprivate IP address of the SIP signaling gateway module 120, the privateIP address of the media gateway module 150 and the private IP address ofthe virtual interface 140. For the SIP signaling message, the SIP-ALGmodule 130 corrects and reconstructs an SDP part such that the SIPsignaling gateway module 120 processes the SIP signaling message. For anRTP packet including voice data, the SIP-ALG module 130 corrects andreconstructs the SDP part such that the media gateway module 150processes the RTP packet.

That is, the SIP-ALG module 130 acquires the unique private IP addressof the SIP signaling gateway module 120 and the unique private IPaddress of the media gateway module 150, corrects and processes theaddress information of the SDP part of the SIP signaling message basedon the private IP address (second address) of the SIP signaling gatewaymodule 120, and modifies and processes the SDP part of the RTP packetbased on the private IP address (third address) of the media gatewaymodule 150.

FIG. 5 is a block diagram illustrating a function of the SIP-ALG moduleof the integrated Internet telephony system 100 shown in FIG. 4.

Referring to FIG. 5, the integrated Internet telephony system 100includes the SIP signaling gateway module 120, the media gateway module150, the SIP-ALG module 130, the LAN interface 110, the WAN interface160 and the virtual interface 140. The SIP signaling gateway module 120is connected to the LAN interface 110 to transmit/receive a SIPsignaling message, and the media gateway module 150 is connected to thevirtual interface 140 to transmit/receive an RTP packet.

Like the construction shown in FIG. 4, the integrated Internet telephonysystem 100 includes three interfaces in a network device layer, thethree interfaces corresponding to a LAN interface 110, a WAN interface160 and a virtual interface 140. IP addresses (e.g., 10.0.0.1,61.77.100.20 and 10.0.1.1) used by the respective interfaces arerespectively allocated.

The integrated Internet telephony system 100 further includes a userinterface 170, such that network operator can set a network policythrough the user interface 170. The network policy is stored in aSIP-ALG Config 171, and is then sent from the SIP-ALG Config 171 to aSIP-ALG management module 132.

The SIP-ALG module 130 includes a SIP-ALG kernel module 131, the SIP-ALGmanagement module 132, a SIP parser 133, a field modification module134, an SDP processing module 135 and a SIP mapping table 136.

The SIP-ALG kernel module 131 binds a SIP signaling message through theLAN interface 110 and the WAN interface 160.

The SIP-ALG management module 132 performs ALG processing on the SIPsignaling message, bound by the SIP-ALG kernel module 131, using the SIPparser 133, the field modification module 134, the SDP processing module135 and the SIP mapping table 136.

Firstly, the SIP parser 133 acts to parse the SIP signaling message thatthe SIP-ALG module 130 received.

For example, it is determined whether or not the SIP signaling message,inputted for parsing by the SIP parser 133, is inbound or outbound, andan IP address, port information and the like are extracted from a headerfield or a payload of the SIP signaling message.

The field modification module 134 functions to modify header fieldvalues of the SIP signaling message using the information of the SIPmapping table 136.

In particular, the field modification module 134 modifies a private IPaddress into a public IP address in a source IP address field or adestination IP address field of an IP header of the SIP signalingmessage.

Further, the field modification module 134 performs a reverse functionto modify the public IP address to the private IP address, and functionsto modify the port field value of the IP header of the SIP signalingmessage.

The SDP processing module 135 modifies values, included in the SDP partof the SIP signaling message, using the information of the SIP mappingtable 136. The modification of the values in the SDP part is also forthe purpose of transmitting/receiving the SIP signaling message betweenthe external SIP terminal and the internal SIP terminal.

The SDP processing module 135 modifies (corrects) the IP addresses ofthe internal and external SIP terminals, included in the SDP part of theSIP signaling message, by mapping IP addresses, respectively, byreferring to the SIP mapping table 136.

A detailed description of the modification of address information on theSDP part of the SIP signaling message will be described later.

The SIP mapping table 136 includes network address translation (NAT)information for the mapping of an inbound or outbound packet.

The information in the SIP mapping table 136 includes addresses andports of external SIP terminals and addresses and ports of internal SIPterminals. The structure of the SIP mapping table 136 will be describedlater.

FIG. 6 is a flow diagram illustrating a method of processing SIPsignaling messages in the integrated Internet telephony system accordingto an embodiment of the invention.

Referring to FIG. 6, the SIP ALG management module 132 binds a SIPsignaling message received by the 5060 port, which is a port fortransmitting/receiving a typical SIP signaling message (S100).

The SIP-ALG management module 132 sends the bound SIP signaling messageto the SIP parser 133, such that the SIP parser 133 parses the SIPsignaling message, particularly, the syntax of the SIP signaling message(S101).

The SIP parser 133 extracts a source address field, a destinationaddress field and port values from the SIP signaling message, and sendsthe extracted fields and values to the SIP-ALG management module 132(S102).

The SIP-ALG management module 132 sends the information of the parsedSIP signaling message to the SDP processing module 135 (S103), and theSDP processing module 135 corrects the SDP part by referring to the SIPmapping table 136 (S104, S105).

The SDP processing module 135 transmits the SIP signaling messageincluding the corrected SDP part to the SIP-ALG management module 132(S106).

The SIP-ALG management module 132 transmits the information, parsed fromthe bound SIP signaling message, to the field modification module 134(S107), the field modification module 134 corrects field values byreferring to the SIP mapping table 136 (S108, S109) and transmits themodified field values to the SIP-ALG management module 132 (S110).

According to the method of the bound SIP signaling message, the SIP-ALGmanagement module 132 causes the SDP part values of the same SIPsignaling message to be corrected (S111, S112, S113 and S114).

The SIP-ALG management module 132 re-injects the SIP signaling message,with the SDP part thereof corrected in the SDP processing module 135 andthe field values thereof modified in the field modification module 134,and particularly, transmits the SIP signaling message to the internal orexternal SIP agent (S115).

The SDP processing module 135 corrects address information on a callsession, set to the SDP part of the SIP signaling message, by referringto the SIP mapping table 136, such that the external and internal SIPagents can transmit/receive an RTP packet through the actual callsession. The field modification module 134 can correct the sourceaddress information and the destination address information of the IPheader of the SIP signaling message into a mapping private or public IPaddress.

If the bound SIP signaling message is a SIP signaling message related tothe initial call session setup, the SDP processing module 135 addsinformation on a transmitting/receiving SIP terminal existing in the SDPpart of the SIP signaling message to the SIP mapping table 136. That is,the SDP processing module 135 allocates a NAT entry to the SIP mappingtable 136 at a time when a new call session is set up, and cancels theNAT entry at a time when the call session is terminated.

This method of managing the SIP mapping table 136 according to thesetting-up and the cancellation of the call session is a method thatdynamically manages the SIP mapping table 136 in order to reinforce thesecurity of the private VoIP network.

While the SIP mapping table 136 of the invention can be staticallymanaged, it is more preferable to dynamically manage the SIP mappingtable 136 since static management is comparable to forming a pin-hole inthe firewall, which promotes information leakage.

Further, the SDP processing module 135 can update NAT information byallocating or cancelling an entry of the SIP mapping table 136 through apseudo character device.

The SDP processing module 135 can also perform an aging function on theNAT information using a contract destroy callback function.

The SDP processing module 135 and the field modification module 134modify the private IP address and port of the SIP signaling message intoa public IP address and port or vice versa using the NAT information ofthe SIP mapping table 136.

For example, the field modification module 134 modifies the private IPaddress and SIP port information of a source terminal, in a header of anoutgoing SIP signaling message (from inside to outside), into a publicIP address and SIP port information, and also modifies the public IPaddress and SIP port of a destination terminal, in a header of anincoming SIP signaling message (from outside to inside), into theprivate IP address and SIP port of an internal SIP agent of thedestination terminal.

The SDP processing module 135 modifies the private IP address and SIPport information of a source terminal, included in an SDP part of anoutgoing SIP signaling message, into a public IP address and SIP port,and also modifies the public IP address and SIP portion information of adestination terminal, included in an SDP part of an incoming SIPsignaling message, into the private IP address and SIP port of thedestination terminal.

Accordingly, the internal SIP terminal and the external SIP terminalexchange the RTP packet including voice data through a call session,based on the address information (IP address and port information)included in the SDP part of the signaling message.

FIGS. 7A and 7B are SIP mapping tables of the integrated Internettelephony system according to an embodiment of the invention, in whichFIG. 7A shows an inbound mapping table and FIG. 7B shows an outboundmapping table.

As shown in FIGS. 7A and 7B, each of the inbound mapping table and theoutbound mapping table can be composed of a validity field, an IPaddress field and a port information field.

As seen from FIG. 7A, the syntax “# iptables-t nat-A PREROUTING-p udp-d211.217.127.38—dport 40000:50000-j SIP_ALG—dir=inbound” defines that NATbe performed on an RTP (UDP) packet based on inbound table information,in which the RTP (UDP) packet is received with a destination port from40000 to 50000 of the external IP address 211.217.127.38 of theintegrated Internet telephony system.

Here, the external IP address of the integrated Internet telephonysystem 100 having the SIP mapping table 136 of FIGS. 7A and 7B is givento be 211.217.127.38.

After the session is set up through the syntax above, when an RTP packetfrom outside is received through the UDP 40000 port of the address211.217.127.38, the integrated Internet telephony system 100 searches anentry with an IP address 211.217.127.38 and port information 4000. InFIG. 7A, as the result of the search, it is assumed that an entry in thesecond row be mapped.

Based on the information of the searched mapping table entry, theintegrated Internet telephony system 100 modifies the destination IPaddress of the RTP packet from 211.217.127.38 into 192.168.0.3, and theport value from 40000 into 30050. The RTP packet modified as such issent to the internal SIP terminal through the VoIP network.

Conversely, in the case of FIG. 7B, the syntax “# iptables-t nat-APREROUTING-p udp-d 192.168.0.1—dport 30000:40000-j SIP_ALG—dir=outbound”defines that NAT is performed on an RTP packet based on outbound tableinformation, in which the RTP packet is received by a port from 30000 to40000 of the LAN IP address 192.168.0.1 of the integrated Internettelephony system 100.

In this set-up state, when an internal SIP terminal having a private IPaddress 192.168.0.100 transmits an RTP packet through a UDP 3000 port,the default gateway address of the internal SIP terminal of theintegrated Internet telephony system 100 is 192.168.0.1, and thus theRTP packet is routed based on outbound table information of theintegrated Internet telephony system 100.

The integrated Internet telephony system 100 searches for an entry, inwhich a source terminal has IP address 192.168.0.100 and portinformation 30000. In FIG. 7B, it is assumed that the entry existing inthe second row of the outbound mapping table be mapped as the result ofthe searching.

Based on the searched information of the mapping table entry, theintegrated Internet telephony system 100 modifies the source IP addressof the RTP packet from 192.168.0.100 to 64.3.2.1, and the port valuefrom 30000 to 60000. The RTP packet modified as above is sent throughthe Internet to the external SIP terminal.

That is, the SIP-ALG module 130 of the integrated Internet telephonysystem 100 modifies the SDP part and header field values of the outgoingSIP signaling message and the in-coming SIP signaling message, such thatthe SIP signaling gateway module 120 processes the SIP signalingmessages. At the same time, the SIP-ALG module 130 manages the outboundIP address and port information and the inbound IP address and portinformation of the RTP packets including voice data using the outboundmapping table and the inbound mapping table, in which the RTP packetsare exchanged through the call session that is set up through the SIPsignaling messages.

The SIP-ALG module 130 modifies the IP address and port information ofthe RTP packets exchanged through the call session, such that the RTPpackets are processed by the media gateway module 150 and thus areactually exchanged between the SIP terminals.

FIG. 8 illustrates a method in which the SIP-ALG module of the inventionmanages the outbound mapping table and the inbound mapping table.

Referring to FIG. 8, the SIP-ALG module 130 manages IP address and portinformation of an internal SIP terminal, included in an SDP part of anoutgoing SIP signaling message, while processing an SDP part of a SIPsignaling message bound at a kernel. The SIP-ALG module 130 also managesIP address and port information of an external SIP terminal through amapping table while processing an SDP part of an incoming SIP signalingmessage.

That is, the SIP-ALG module 130 extracts IP address and port informationof internal and external SIP terminals from the SDP part of the SIPsignaling message and manages the extracted IP address and portinformation.

As seen from FIGS. 7A and 7B, the SIP-ALG module 130 corrects IP addressand port information of an RTP packet, received from the internal SIPterminal, by referring to the outbound mapping table, and corrects IPaddress and port information of an RTP packet received from the externalSIP terminal by referring to an inbound mapping table.

The SIP-ALG module 130 dynamically manages the SIP mapping table 136 byallocating/cancelling an entry to/from the SIP mapping table 136 througha pseudo character device. For example, the SIP-ALG module 130 canallocate/delete or write/read an entry value of the SIP mapping table136 through a write function and an input-output control (IOCTL)function.

Hereinafter, flows of a SIP signaling message in the integrated Internettelephony system of the invention will be described by way of example.

FIGS. 9A to 9F are conceptual views illustrating SIP signaling messageflows in the integrated Internet telephony system according to anembodiment of the invention.

FIG. 9A shows a message flow in which a SIP signaling message istransmitted from the internal SIP terminal to the external SIP terminalof the private VoIP network. As shown in FIG. 9A, the internal SIPterminal generates the SIP signaling message (e.g., a call requestmessage), in which the destination address information in the headerarea of the SIP signaling message is set to the IP address and portinformation (e.g., 6060) of the external SIP terminal, and the sourceaddress information in the header area of the SIP signaling message isset to the address information and port information of the internal SIPterminal, and the private IP address of the internal SIP terminal isincluded in a payload field.

The LAN interface 110 receives the SIP signaling message from theinternal terminal and forwards the received SIP signaling message to theSIP-ALG module 130. A block of the SIP-ALG module 130, managing anoutbound SIP signaling message, modifies the destination addressinformation to the IP address information of the LAN interface 110(PRE-ROUTING) and to the IP address information (public IP address) ofthe WAN interface 160 of the integrated Internet telephony system 100,and modifies the IP address of the internal SIP terminal in the payloadfield to the external IP address (public IP address).

The SIP-ALG module 130 modifies the internal IP address in the payloadfield to correspond to the external IP address while modifying thedestination and source address information of the SIP signaling message,received from the internal SIP terminal, based on the SIP mapping table136, such that the internal SIP terminal and the external SIP terminalcan set the destination and source address information of their own RTPpackets and exchange the set destination and source address informationwith each other.

FIG. 9B shows a message flow in which a SIP signaling message istransmitted from the external SIP terminal to the internal SIP terminal.As shown in FIG. 9B, the external SIP terminal generates the SIPsignaling message (e.g., a call request message), in which destinationaddress information in the header area of the SIP signaling message isset with the public IP address and port information (e.g., 5060) of theWAN interface 160 of the integrated Internet telephony system 100, andsource address information is set with the IP address information andport information of the external SIP terminal.

The external SIP terminal sets the public IP address of the WANinterface 160, which is the public IP address of the internal SIPterminal, as the destination address information since it is notnotified of the internal IP address (private IP address) of the internalSIP terminal.

The WAN interface 160 receives the SIP signaling message from theexternal SIP terminal and sends the received SIP signaling message to ablock of the SIP-ALG module 130, which manages inbound SIP signalingmessages. The SIP-ALG module 130 modifies the destination addressinformation from the IP address information of the WAN interface 160 tothe IP address information of the LAN interface 110 (i.e., from thepublic IP address to the private IP address), modifies the sourceaddress information from the address information of the external SIPterminal to the IP address information of the LAN interface 110, andtransmits the SIP signaling message with the modified source anddestination address information to the internal SIP terminal(PRE-ROUTING).

That is, the SIP-ALG module 130 modifies the destination and sourceaddress information of the SIP signaling message received from theexternal SIP terminal based on the SIP mapping table 136, such that theinternal SIP terminal and the external SIP terminal can set thedestination and source address information of their own RTP packets andexchange the set destination and source address information with eachother.

FIG. 9C shows a message flow in which the internal SIP terminaltransmits a SIP response message in response to an SIP signaling messagereceived from the external SIP terminal. As shown in FIG. 9C, theinternal SIP terminal transmits the SIP response message (e.g., a callresponse message) in response to the SIP signaling message (e.g., a callrequest message) received from the external SIP terminal. Here, thedestination address information of the SIP response message is set withthe private IP address and port information of the LAN interface 110,the source information is set with the IP address information and portinformation of the internal SIP terminal, and a payload field includesthe internal IP address of the internal SIP terminal (i.e., the privateIP address of the internal SIP terminal).

The SIP-ALG module 130 modifies the destination address information ofthe SIP response message, forwarded through the LAN interface 110, tothe IP address information and port information of the external SIPterminal, extracted from the header of the SIP signaling messagereceived from the external SIP terminal. The SIP-ALG module 130 alsomodifies the source address information to the IP address and portinformation of the WAN interface 160, and modifies the internal IPaddress in the payload field into an external IP address, so as totransmit the forwarded SIP response message to the external SIP terminalthrough the WAN interface 160.

Accordingly, the internal SIP terminal and the external SIP terminal canexchange RTP packets through the integrated Internet telephony system100. In addition, the internal SIP terminal and the external SIPterminal can substantially exchange RTP packets by setting thedestination address and source address of the RTP packets with theaddress information extracted from the header of the SIP signalingmessage and the address information included in the payload.

FIG. 9D shows a message flow in which the SIP signaling gateway module120 of the integrated Internet telephony system 100 transmits a SIPresponse message to the external SIP terminal, FIG. 9E shows a messageflow in which a SIP signaling message is transmitted from the externalSIP terminal to the SIP signaling gateway module 120, and FIG. 9F showsa message flow in which the SIP signaling gateway module 120 transmits aSIP response message in response to the SIP signaling message receivedfrom the external SIP terminal.

Referring to FIGS. 9D to 9F, the SIP signaling gateway module 120 isconnected to the LAN interface 110, and transmits the SIP signalingmessage by setting the source address information of the SIP signalingmessage to the address information of the LAN interface 110, setting theaddress information of the external SIP terminal in the destinationaddress information, and including the internal IP address informationin a payload field.

The SIP-ALG module 130 modifies the internal IP address in the payloadwith the external IP address while modifying the source addressinformation of the SIP signaling message, received from the SIPsignaling gateway module 120, with the address information of the WANinterface 160.

In the meantime, the SIP-ALG module 130 modifies the source addressinformation of the SIP signaling message, received from the external SIPterminal, to the address information of the LAN interface 110 so as tobe received by the SIP signaling gateway module 120.

Further, the SIP-ALG module 130 modifies the destination addressinformation of the SIP response message, generated by the SIP signalinggateway module 120, from the address information of the LAN interface110 to the external IP address and port information extracted from theheader of the SIP signaling message, received from the external SIPterminal, and modifies the internal IP address included in the payloadto the external IP address before transmitting the SIP response messagethrough the WAN interface 160.

Hereinafter, a description will be given of an address processing methodof the integrated Internet telephony system 100 according to anembodiment of the invention.

Firstly, in the integrated Internet telephony system 100, the SIP-ALGfunction, the SIP signaling gateway function and the media gatewayfunction are integrated, such that the SIP signaling gateway module 120uses the second address, which is aliased from the primary IP address(the first address) that the SIP-ALG module 130 uses.

The SIP-ALG module 130 binds the SIP signaling message to the firstaddress (the primary IP address), and the SIP signaling gateway module120 binds the SIP signaling message to the aliased second address.

In the integrated Internet telephony system 100, the virtual interface140 is also provided in addition to the LAN interface 110 connected tothe private VoIP network (the internal network) and the WAN interface160 connected to the Internet (the external network).

The private IP address of the virtual interface 140 can be set based onthe private IP address of the LAN interface 110. For example, when theprivate IP address of the LAN interface 110 is 10.0.0.1, the private IPaddress of the virtual interface 140 can be set to 10.0.1.1.

The integrated Internet telephony system 100 allows the media gatewaymodule 150 to use the third address aliased from the private IP addressof the virtual interface 140.

The SIP-ALG module 130 of the integrated Internet telephony system 100acquires the second address (e.g., 10.0.0.2) that the SIP signalinggateway module 120 uses and the third address (e.g., 10.0.1.2) that themedia gateway module 150 uses.

When the SIP signaling message is bound at the kernel, the SIP-ALGmodule 130 corrects the SDP part while modifying the address and portinformation of the SIP signaling message, such that the SIP signalinggateway module 120 can process the SIP signaling message.

The SIP-ALG module 130 extracts a source address field, a destinationaddress field and port values by parsing the syntax of the SIP signalingmessage, which is bound at the kernel, and corrects the SDP part byreferring to the SIP mapping table 136.

As seen from FIGS. 9A to 9F, the SIP-ALG module 130 modifies andcorrects the address and port information and the SDP part of the boundSIP signaling message, which is bound at the kernel, by referring to theSIP mapping table 136.

Accordingly, the SIP-ALG module 130 transmits the SIP signaling messageto the SIP signaling gateway module 120 by modifying the address andport information of the message, such that the SIP signaling gatewaymodule 120 can set a call session by processing the SIP signalingmessage.

The integrated Internet telephony system 100 manages the IP address andport information of the internal or external SIP terminal (SIP agent),included in the SIP signaling message, using the SIP mapping table 136.

As illustrated with reference to FIG. 8, the SIP-ALG module 130 of theintegrated Internet telephony system 100 processes the SDP part of theSIP signaling message, which is bound at the kernel, manages the IPaddress and port information of the internal SIP terminal included inthe SDP part of the outgoing SIP signaling message, processes the SDPpart of the incoming SIP signaling message, and manages the IP addressand port information of the external SIP terminal through the outboundmapping table.

When the call session is set between the internal SIP terminal and theexternal SIP terminal, the integrated Internet telephony system 100modifies the address information (the IP address and port information)of the destination terminal, included in the SDP part of the SIPsignaling message, such that RTP packets including voice data can beexchanged between the internal and external SIP terminals.

Accordingly, the internal and external SIP terminals can exchange theRTP packets based on the address information of the destination terminalincluded in the SDP part of the SIP signaling message.

Although the present disclosure has been described with an exemplaryembodiment, various changes and modifications may be suggested to oneskilled in the art. It is intended that the present disclosure encompasssuch changes and modifications as fall within the scope of the appendedclaims.

1. A voice of Internet protocol (VoIP) network comprising: an integratedInternet telephony system having an application layer gateway function,a signaling processing function and a media processing functionintegrated in one architecture, wherein the integrated Internettelephony system performs an application layer gateway (ALG) processingon a packet bound to a first address and processes a signaling messagebound to a second address so as to set up a call session between anexternal session initiation protocol (SIP) terminal connected to apublic network and an internal SIP terminal connected to a privatenetwork, and performs a media processing on a real time protocol (RTP)packet based on a third address, the RTP packet exchanged through thecall session.
 2. The VoIP network according to claim 1, wherein theintegrated Internet telephony system comprises: a first interfaceconnected to the public network; a second interface connected to theprivate network; and a third interface connected to a different networkfrom the second interface.
 3. The VoIP network according to claim 1,wherein: the first address is a private Internet protocol (IP) addressof the second interface, the second address is in a same network as thefirst address, but is a different logical IP address from the firstaddress, and the third address is in a same network as the thirdinterface, but is a different logical IP address from the thirdinterface.
 4. The VoIP network according to claim 3, wherein: the secondaddress is an IP address aliased from the first address; and the thirdaddress is an IP address aliased from an address of the third interface.5. The VoIP network according to claim 1, wherein the integratedInternet telephony system modifies a private or public addressinformation in a header field of the signaling message, which is boundto the first address, by mapping the public or private addressinformation.
 6. The VoIP network according to claim 5, wherein theintegrated Internet telephony system modifies a destination addressinformation of the signaling message, which is bound to the firstaddress, to the second address.
 7. The VoIP network according to claim1, wherein the integrated Internet telephony system manages address aninformation of the internal SIP terminal, included in a sessiondescription protocol (SDP) part of the signaling message, using aninbound mapping table when the signaling message is a outgoing from theprivate network to the public network, and manages address informationof the external SIP terminal included in the SDP part of the signalingmessage using an outbound mapping table when the signaling message is anincoming message from the public network to the_private network.
 8. TheVoIP network according to claim 7, wherein the integrated Internettelephony system modifies a destination address information of an RTPpacket by referring to the inbound mapping table when the RTP packet isan outgoing packet from the public network, and modifies the destinationaddress information of the RTP packet by referring to the outboundmapping table when the RTP packet is an incoming packet from the privatenetwork.
 9. The VoIP network according to claim 1, wherein the internalSIP terminal or the external SIP terminal exchanges the RTP packet basedon the destination address information included in the SDP part of thesignaling message when the call session is set up.
 10. An integratedInternet telephony system, which provides an Internet telephone servicethrough a voice of Internet protocol (VoIP) network, comprising: a localarea network (LAN) interface connected to a private VoIP network using afirst private Internet protocol (IP) address; a wide area network (WAN)interface connected to a public network using a public IP address; avirtual interface having a virtual private IP address; a sessioninitiation protocol (SIP) signaling gateway module processing asignaling message, which is bound to a second private IP address, so asto set up a call session; a media gateway module connected to thevirtual interface through a third private IP address so as to process areal time protocol (RTP) packet; and a session initiationprotocol-application layer gateway (SIP-ALG) module modifying adestination address information of the bound signaling message andaddress information included in a session description protocol (SDP)part of the bound signaling message.
 11. The integrated Internettelephony system according to claim 10, wherein the SIP-ALG modulemodifies a destination address information of an outgoing signalingmessage and a private address information included in an SDP part of theoutgoing signaling message to a public address information.
 12. Theintegrated Internet telephony system according to claim 10, wherein theSIP-ALG module modifies a destination address information of anin-coming signaling message to the first private IP address, andmodifies a public address information of an external SIP terminalincluded in the SDP part to a private address information.
 13. Theintegrated Internet telephony system according to claim 10, wherein theSIP-ALG module includes: a SIP parser parsing the bound signalingmessage; a field modification module modifying an address information inan IP header of the signaling message; an SDP processing moduleconverting an address information in the SDP part of the signalingmessage; and a SIP mapping table storing an address information mappedto an address information included in the signaling message when thesignaling message is an incoming or an outgoing message.
 14. Theintegrated Internet telephony system according to claim 10, wherein: thesecond private IP address is an IP address aliased form the firstprivate IP address, and the third private IP address is an IP addressaliased from the virtual private IP address.
 15. The integrated Internettelephony system according to claim 10, wherein the SIP-ALG modulemodifies a destination address information of the signaling message tothe first private IP address when the signaling message is an incomingmessage, and modifies a source address information of the incomingsignaling message to the first private IP address.
 16. The integratedInternet telephony system according to claim 15, wherein the SIP-ALGmodule sets the public IP address as a destination address informationin the SDP part of the signaling message.
 17. The integrated Internettelephony system according to claim 10, wherein the SIP-ALG modulemodifies a source address in formation of the signaling message to thepublic IP address when the signaling message is an outgoing message, andmodifies an address information in the SDP part to an external IPaddress of an external terminal.
 18. A signaling method of an integratedInternet telephony system, which has a session initiationprotocol-application layer gateway (SIP-ALG) function, a sessioninitiation protocol (SIP) gateway unction and a media processingfunction integrated in one architecture, and which includes a firstinterface connected to a private voice of Internet protocol (VoIP)network and a second in interface connected to a public network, themethod comprising: adding, at the integrated Internet telephony system,a virtual interface belong to a different network from a first address;managing, at a first module in charge of the SIP-ALG function, anaddress information included in a signaling message, which is bound tothe first address, using a mapping table; modifying, at the firstmodule, the address information of the signaling message and an addressinformation included in a session description protocol (SDP) part of thesignaling message; setting up, at a second module in charge of the SIPsignaling gateway function, a call session based on the signalingmessage, which is bound to a second address; and exchanging, at aninternal SIP terminal in the private VoIP network and an externalterminal in the public network, a real time protocol (RTP) packet usinga third address.
 19. The signaling method according to claim 18, furthercomprising: performing, at a third module in charge of the mediaprocessing function, a media processing on the RTP packet using thethird address, which is connected to the virtual interface.
 20. Thesignaling method according to claim 18, wherein the second address is anallocated address, which is aliased from the first address.
 21. Thesignaling method according to claim 19, wherein the third address is anallocated address, which is aliased form the IP address of the virtualinterface.
 22. The signaling method according to claim 19, whereinmanaging, at the first module in charge of the SIP-ALG function, anaddress information included in a signaling message, which is bound tothe first address, using a mapping table, comprises: managing a privateaddress information of the internal SIP terminal in the SDP part of thesignaling message, which is going out from the private VoIP network tothe public network, using an inbound mapping table; and managing apublic address information of the external SIP terminal in the SDP partof the signaling message, which is coming in from the public network,using an outbound mapping table.
 23. The signaling method according toclaim 22, wherein modifying, at the first module, an address informationof the signaling message and an address information included in an SDPof the signaling message, comprises: modifying the address informationof the signaling message to correspond to a private IP addressinformation or a public IP address information by referring to arespective one of the mapping tables.
 24. The signaling method accordingto claim 23, wherein modifying, at the first module, an addressinformation of the signaling message and an address information includedin an SDP of the signaling message, comprises: modifying a destinationsource address information and a source address information of thesignaling message, which is coming in, from the public IP address to thefirst private IP address and from an address information of the externalSIP terminal to the first private IP address by referring to the inboundmapping table, and transmitting the signaling message to the internalSIP terminal.
 25. The signaling method according to claim 23, whereinmodifying, at the first module, an address information of the signalingmessage and an address information included an SDP of the signalingmessage, comprises: modifying a source address information of the SIPsignaling message and a source information in a payload field of the SIPsignaling message, which is going out, to the public IP addressinformation and an external IP address of the external SIP terminal byreferring to the outbound mapping table.
 26. The VoIP network accordingto claim 2, wherein: the first address is a private Internet protocol(IP) address of the second interface, the second address is in a samenetwork as the first address but is a different logical IP address fromthe first address, and the third address is in a same network as thethird interface, but is a different logical IP address from the thirdinterface.